The auditory system encodes sound by decomposing the amplitude signal arriving at the ear into multiple frequency bands whose center frequencies and bandwidths are approximately exponential functions of the distance from the stapes. This organization is thought to result from the adaptation of cochlear mechanisms to the animal's auditory environment. Here we report that several basic auditory nerve fiber tuning properties can be accounted for by adapting a population of filter shapes to encode natural sounds efficiently. The form of the code depends on sound class, resembling a Fourier transformation when optimized for animal vocalizations and a wavelet transformation when optimized for non-biological environmental sounds. Only for the combined set does the optimal code follow scaling characteristics of physiological data. These results suggest that auditory nerve fibers encode a broad set of natural sounds in a manner consistent with information theoretic principles.

/pdf/efficient-coding-of-natural-sounds-52z91f3zkh.pdf

Efficient coding of natural sounds.

The speech encoding apparatus (220) has a first layer encoding section (2201), a first layer decoding section (2202), a delay section (2203), a subtracting section (104), a frequency domain transforming section (101), a second layer encoding section (105) and a multiplexing section (106).

Encoding device, decoding device, and method thereof

A scalable decoding apparatus capable of providing decoded audio signals of high quality having less degradation of a high frequency spectrum even when decoding audio signals by generating the high frequency spectrum by use of a low frequency spectrum. In the apparatus, an amplitude adjusting part (1211) uses different adjustment coefficients in accordance with the characteristic of first layer spectrum information to adjust the amplitude of a first layer decoded signal spectrum, and then outputs the amplitude-adjusted first layer decoded signal spectrum to a pseudo-spectrum generating part (1012). Using amplitude-adjusted first layer decoded signal spectrum received from the amplitude adjusting part (1211), the pseudo-spectrum generating part (1012) generates and outputs a pseudo-spectrum of high frequencies to a scaling part (1013). The scaling part (1013) scales the spectrum received from the pseudo-spectrum generating part (1012) and then outputs it to an adder (B).

Scalable decoding apparatus and scalable encoding apparatus

An overall encoding procedure and associated decoding procedure are presented. The encoding procedure involves at least two signal encoding processes operating on signal representations of a set of audio input channels. Local synthesis is used in connection with a first encoding process to generate a locally decoded signal, including a representation of the encoding error of the first encoding process. This locally decoded signal is applied as input to a second encoding process. The overall encoding procedure generates at least two residual encoding error signals from at least one of said encoding processes, including at least said second encoding process. The residual error signals are then subjected to compound residual encoding in a further encoding process, preferably based on correlation between the residual error signals.

Joint enhancement of multi-channel audio

The invention relates to the technical field of audio encoding and/or decoding technologies, and thus concerns an overall encoding procedure and associated decoding procedure. The encoding procedure involves at least two signal encoding processes (S1-S3) operating on signal representations of a set of audio input channels, as well as residual encoding (S7-S8). It also involves a dedicated process (S4-S6) to estimate and encode energies of the audio input channels. Each encoding process is associated with a corresponding decoding process. In the overall decoding procedure the decoded signals from each encoding process are preferably combined such that the output channels are close to the input channels in terms of energy and/or quality. Normally, the combination step also adapts to the possible loss of one or more signal representation in part or in whole, such that the energy and quality is optimized with the signals at hand in the decoder. In this way, the overall quality of the output channels is improved.

Energy conservative multi-channel audio coding

Networks are getting more and more heterogeneous. Scalable codecs are especially suited for such a context as they permit the bitrate to be lowered in a simple way, at any point of the transmission, for adaptation to network conditions and to terminal capacities. Classically, scalable codecs are organised in layers and scalability is obtained by sending more or fewer layers to the decoder. The obtained granularity depends on the layer sizes, and the available bitrates are fixed and limited in number. The paper presents a novel scalable audio coding scheme where the bitrates vary continuously between a minimal and a maximal value, allowing free modification of the bitrate. With this novel approach, all bitrates are valid, sending even one more bit results in different output signal with statistically growing quality. Test results show that this method provides quality as good as or even better than that of a non-scalable version.

A scalable speech and audio coding scheme with continuous bitrate flexibility

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A scalable speech and audio coding scheme with continuous bitrate flexibility